Low-Latency Telephony Pipeline

Voice AI Agents
Conversational VoIP Engineering.

Automate inbound call centers and outbound customer dispatch over PSTN and SIP trunks. We build voice pipelines targeting sub-450ms turnaround latency using Deepgram STT, fast LLM reasoning, and Cartesia/ElevenLabs streaming speech synthesis.

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<450ms

Audio Pipeline Latency

Sub-second conversational pacing eliminating awkward pauses over PSTN/SIP channels.

Multi-Tenant

Concurrent Telephony

Process thousands of parallel inbound phone sessions without line saturation or busy signals.

SIP Trunk

Warm Call Handovers

Instant SIP transfers to human call center queues with full live transcript context.
Capabilities

Technical Capabilities

WebRTC & SIP Audio Streaming

Minimize round-trip audio latency. We construct bidirectional WebSockets and SIP media gateways that stream caller audio directly into real-time speech-to-text models.

Real-Time VAD & Barge-In Interruption

Human speech is non-linear. Our Voice Activity Detection (VAD) models instantly pause AI audio playback when a user interrupts, updating conversation states without dropping context.

Streaming Text-to-Speech (TTS)

Integrate streaming audio synthesis engines (Cartesia Sonic, ElevenLabs). We stream PCM audio chunks back to the phone line as tokens are generated.

PBX & Call Center Integrations

Native integration with Twilio, Plivo, Genesys, Asterisk, and Cisco Unified Communications for automated call dispatching, IVR deflection, and appointment scheduling.

Architecture

Enterprise Voice Stack

We orchestrate low-latency VoIP WebSockets, deep learning transcript models, and resilient scale containers.

Telephony Hooks

Handles inbound VoIP calls, WebRTC audio streams, and SIP call routing dynamically.

Twilio Voice Vapi Platforms Retell AI

Speech-to-Text

Processes audio streams in real-time, outputting highly accurate transcripts in under 100ms.

Deepgram Nova-2 Whisper Models WebRTC Buffers

Text-to-Speech

Generates life-like voice responses with custom intonations and sub-second execution intervals.

ElevenLabs TTS Cartesia Sonic Interruption VAD

Hosting Pipelines

Streams call queues dynamically, maintaining server execution limits via container scaling policies.

AWS ECS Fargate Redis Queue WebRTC Sync
Voice Security & Telephony Governance

Sub-450ms Telephony & Streaming Audio Stack

Low-latency PSTN/SIP WebSockets, Deepgram STT, ElevenLabs TTS, and real-time VAD barge-in handling.

Sub-450ms VoIP Latency SLA

Optimized WebSockets streaming pipelines deliver sub-450ms end-to-end voice latency for natural, human-grade conversations.

<450ms Latency WebSockets

VAD Barge-In Interrupts

Voice Activity Detection (VAD) algorithms instantly pause TTS audio playback the moment the caller speaks mid-sentence.

VAD Barge-In Zero Delay

PSTN & SIP Trunk Security

Direct integration with Twilio Voice, Plivo, and enterprise PBX SIP trunks featuring TLS audio payload encryption.

SIP Encryption Twilio Voice

Deepgram STT & Cartesia TTS

Streaming speech-to-text and neural voice synthesis fine-tuned for industry accents and noisy background calls.

Deepgram Nova Cartesia TTS
Engineering Lifecycle

Voice AI Telephony Implementation Pipeline

From SIP trunk configuration to ultra-low latency voice agent deployment in 30 days.

01 Days 1–5

SIP & Call Flow Audit

Audit call center IVR flows, configure SIP trunking credentials, map voice intents, and select custom TTS voices.

02 Weeks 2–3

Streaming Stack Build

Wire Deepgram STT $\rightarrow$ Groq LLM $\rightarrow$ Cartesia TTS streaming WebSockets pipeline with VAD barge-in controls.

03 Week 4

Latency & Noise Hardening

Conduct acoustic noise testing, optimize audio buffer packet sizes, and tune sub-450ms response thresholds.

04 Day 30+

Voice Agent Call Launch

Deploy scalable VoIP agents handling thousands of concurrent inbound/outbound call sessions.

FAQ

Common Questions

Everything you need to know about our enterprise services.

Our optimized pipelines achieve a response latency under 800ms, making conversations feel natural and preventing awkward silence.

Yes. We configure the telephony system to execute SIP refer actions, transferring callers to your office phone lines or call center queues.

Our voice system uses specialized voice activity detection (VAD) filters that ignore background noise and detect user speech boundaries immediately.

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